Music, Programming, and other topics related to the modern Music Engineering Scene.

Going any deeper into anything audio-related requires some basic understanding of both Capacitors and Inductors. This will definitely be a multi-part series, as there's a lot to cover. But fear not! After all's said and done, we'll be able to dive into the really interesting parts of audio electronics and we'll be reading schematics in no time.

Before going further, it's important to note: capacitors and inductors are reactive elements. Basically, they only work with AC current. Putting DC through a capacitor makes it act like an open circuit, while DC through an inductor makes it act like a short circuit. This can be helpful in some cases, but we'll explore those later on.

We briefly touched on the capacitor when talking about condenser microphones, but we need to go into a little more detail on how this thing actually works. Consider the following circuit diagram:Basically, we have an input voltage, traveling across a capacitor to the output voltage. The resistor coming down connecting to ground makes this circuit a voltage divider. Basically, the input voltage gets divided between the capacitor and the resistor. These voltage dividers occur quite frequently, so it's best to get familiar with them.

Now, let's pretend we put our AC current in at Vi. What happens when it oscillates at a low frequency? What about at a high frequency? We can't say yet, because we don't quite know how capacitors work. See if you can wrap you mind around this: When a current flows through a capacitor, electrons build up on one of the plates, creating a potential difference. When current stops flowing, the electrons that were built up flow back to where they came from. You can imagine that this is sort of like Alternating Current. If you apply too much current, too many electrons will build up on one side of the capacitor, and they will jump to the other side - a process known as arcing. If your capacitor arcs, it is fried, broken, kaput. You'll have to replace it.

With this conceptual image in mind, lets take a look at the equations. I must put a disclaimer in here before we begin that we journey into the realm of imaginary numbers for this next part. Talking about a capacitor in terms of its capacitance isn't very helpful when dealing with a voltage divider. It's much easier to think of it as an impedance (resistance). To find out what impedance a capacitor has, we use this equation:Zc = 1/(j*w*C)Where Zc is the impedance of the capacitor, j is the square root of -1, w is the oscillating frequency of the input, and C is the capacitance of the capacitor. Don't let the j term bother you. In math, the square root of -1 is usually denoted by the letter i, but since current has the letter i in electrical analysis, we change i to j. w (should be an omega, but I'm not sure how to make the character for it), is the oscillating frequency of the signal. Basically, we have an alternating current. What frequency is it oscillating at? (Outlets in your house oscillate at 60Hz) C is the capacitance, measured if farads. This value is given to you on all capacitors, and goes back to that Q = CV equation with the microphones. Don't worry too much about that, though, as the capacitance is always provided to you by the manufacturer of the capacitor.

Let's take a minute to regroup before going on. Reread the last paragraph a few times, and let's do some number crunching when you're feeling at least half confident about everything.

What happens when we put in low values for w (we have low oscillating frequencies)? Well, 1 divided by something really small comes out really big, right? So our impedance is really high, and nothing really flows through the capacitor. Now let's put in huge value for w (really high oscillating frequencies). One divided by something really big (close to infinity) is really small. So, our capacitor has a low impedance, and signal flows through the capacitor really easy.

Now imagine this: Instead of passing electrical current through that circuit, let's pass an audio signal. Low frequencies won't make it past the capacitor, but higher frequencies will! Now, you can see that this is the circuit for a very simple high-pass filter.

Take a moment to let that sink in, and next time we'll go over inductors and low-pass filters.

If you've read any sort of spec sheet for speakers, you've undoubtedly seen the term dB (SPL) scribbled on there a few times. What is this crazy thing, and why does it matter so much?

dB is short for decibel, which is the unit of loudness. There's some crazy equation involving logarithms that describes it:dB = 20 * log (Vm/Vref)where Vm is the measured value and Vref is the reference value. This is the general decibel equation. This gives you different categories of decibels based on what you make Vref. This is where dBu and dBV come from on spec sheets. (dBu's vref = .775V and dBV's = 1V). SPL simply stands for Sound Pressure Level, which basically specifies you're dealing with audio.

Optimal monitoring levels while mixing are around 85dB, but only for 30-45 minutes at a time. Any longer than that, and you'll risk damaging your ears. It's good to take breaks anyways so you don't get burnt out.

When outside, you have to worry not to get too loud with your dB SPL. Each city has its own laws, but usually, by law, you're not allowed to exceed 95dB SPL in noise outside of your party/club/anything. If someone measures how loud you are and it exceeds the boundaries in your area, you could be slapped with a pretty bad fine, so be sure to watch out for that.

Moral of the story: keep the volume down. It will preserve your ears and keep you out of trouble.

If you have anything remotely resembling a big time studio, you're going to have at least a small rack. If you don't know what a rack is, it's basically a big metal frame that you mount all of your outboard gear on. If you accrue a lot of outboard gear, organizing your rack will be very helpful for anyone who uses your studio.

The first step to having a successful organization scheme is to separate all your gear into different categories. Put all your preamps in one place, power amplifiers in another place, and effects gear in a different place. Organize your effects by category (eg. reverb, dynamics, EQ...). Once you have everything broken down into groups, it's time to find the best rack space for it.

If all you have is one vertical rack, great. Load everything on in a sensible way. For amplifiers, pick a power-on convention and stick with it. The studio I use goes with "turn everything on from the bottom up", so we mounted our amplifiers in order of what has to turn on first to prevent pops and things. Then, we turn them back off from the top down. This is very important, as it will prevent you from forgetting which order to turn things on in and help keep your equipment undamaged.

If you have many racks side-by-side, think of the most efficient grouping method. You definitely want all your effects as close as possible, so maybe use the top half of two racks, whatever works best for you. The point is to have all your effects accessible at one glance, making it easy to pick out which unit you want.

If you have any other miscellaneous gear, find a suitable spot and go with it, just make sure it makes logical sense to put it there. Is this the first place you'd look for this hardware? If not, you might consider moving it.

Lastly, but most importantly, don't make a mess with the wiring. If you spend all this time putting things in the right place, make sure you hook them up neatly. Remember to always keep audio cables separate from electrical cables and try not to get anything tangled so you can easily replace a cable if it goes bad.

If you keep your rack nice and organized, you'll find all your sessions will run smoother and your time spent in the studio will be a lot happier.

No, polar patterns have nothing to do with polar bears traversing the Arctic. With that out of the way, we can begin...

Polar Patterns are used to describe how microphones pickup sound around them. Are they directional? omnidirectional? cardiod? We'll take a look at what each of these means, and dispell a few common misconceptions.

There are three main types of polar patterns: Omnidirectional, Bidirectional, and Directional. Luckily for us, these names fit perfectly how each mic reacts. Omnidirectional mics pickup sound from all directions, bidirectional pick up sounds from two directions, and directional mics pickup sound from one direction. How might each of these work? Let's break it down even smaller.

Omnidirectional mics are often referred to as "Omnis," to save time on stumbling over such a long name. It might seem a little strange that a microphone can pickup sound from all directions, but it is very possible and in fact very easy. Consider a microphone without any "vents" in the side of it. There's simply one opening at the end for sound waves to hit the diaphragm. It's easy to see that sound waves coming from the front of this mic will vibrate the diaphragm, but what about ones coming from the back? These will vibrate the diaphragm, too. Sound waves actually bend, or diffract, around objects. When a sound wave hits the microphone from behind, it bends around the edge of it, into the diaphragm. It may seem a little odd, but it works.

Bidirectional mics, commonly referred to as "Figure-8 mics," pick up sound from two directions. They can do this with either a single diaphragm, or two put back to back. When you talk into one of the sides, the diaphragm will vibrate like normal. However, if you talk to it halfway between each side, you will not get any signal from the mic. The sound waves are equal on each side of the diaphragm, canceling each other out. This is an interesting effect, and helps in reducing bleed from other instruments.

Directional mics come in many types. The most common type is called "cardiod," because a graph of this polar pattern looks like a heart. This type of mic will pickup sound from only one direction. Hypercardiod mics are similar to cardiod, but you can consider them to be halfway between cardiod and bidirectional. They pick up a little sound from behind them, but not much. "Shotgun mics" are cardiods to the extreme. These bad boys are super directional. You can point them at a spot from a pretty good distance away, and they only pick up sounds in that spot.

Keep in mind, directional mics tend to have vent-like slits on the sides of the microphone. If you were to cover up these vents, the mic will become omnidirectional. This can cause feedback/bleed problems, so be sure not to cover the vents. Also, all of the above polar patterns can be found in both dynamic and condenser mics.

One more thing: All directional and bi-directional mics have this crazy thing called Proximity effect. We mentioned it in an earlier blog post, but there was a slight error there that I'm correcting now. Before, I said something like "dynamic mics have proximity effect." While this is true, condenser mics also have proximity effect. As long as a microphone has a directional polar pattern, it will have proximity effect. The more directional a mic is, the more proximity effect it will have.

Polar patterns are wicked cool and can greatly effect your recording. Be sure to choose the right pattern for your situation!

Before going much further, it's very important to understand just how this whole electricity thing works. So I'll give a brief history, draw a few pictures, and hopefully shed some light on the mysteries of the electrical world.

Consider this very simple circuit:There is more voltage at the positive terminal than the negative terminal, and the current (I) of the circuit travels clockwise. That weird squiggly line is just a resistor. This is all well and good, but it is opposite of what physically happens. The above picture is correct from an analytical stand point, but consider the following: Current can be described as the flow of electrons. Electrons are negatively charged. Something with high voltage has less electrons than something with low voltage. Therefore, the actual, physical current flows counter clockwise in the picture above. The reason we describe it the opposite way is because that's how Ben Franklin did it, and he was wrong. He pictured current as the flow of positive charges, or protons. Protons, however, don't move around. Rather than change all the history books upon this discovery, everyone gritted their teeth and said "okay, we know this is opposite of how it works, but let's just refer to current as the flow of positive charge."

So if current is the flow of electrons, what exactly is voltage? Thanks to Ben Franklin, we define voltage as the amount of positive charge something has with respect to a "ground" of some sort. Basically, we pick a point and say "let's call this our reference point and say that this is zero volts." Let's call this point A. Point A has a specific number of electrons. Anything with this number of electrons will also have zero voltage. Now here's where it gets tricky again. Anything with more electrons has a lower voltage, and anything with less electrons has a higher voltage. So you can think of voltage as a point's craving for electrons. The more it wants it electrons, the higher voltage it has. If it has too many electrons, it will have a lower voltage.

Point A is connected to ground, and is our reference voltage. Point B has a higher voltage than A, because of the voltage source (the circle on the left). After the current goes across the resistor, there's a voltage drop, and you get back to your original A voltage.

In the picture above, you can imagine the "-" side of the voltage source pushing out electrons, and the "+" side sucking them up. However, we say the current flows clockwise, because we like to think of current as positive charge, not negative.

So that leaves us with resistance. What exactly is resistance? Resistors are essentially carbon. The more carbon you have, the bigger a resistor you have. There's a little more to it than that, but resistors are essentially carbon wrapped up in plastic. You can make your own "resistor" by scribbling on a piece of paper with a pencil. Then, take a Digital Volt Meter and measure the resistance from one side of your scribble to the other. You might be surprised by what you see... just don't try to use that in an electrical circuit. The paper will burn up with too much voltage/current and then you'll have quite the fire on your hands.

And that's the basics of electrical circuits. Please feel free to comment with questions, as it is a little counter-intuitive, but it all makes sense in the end.

The word "Jitter" to many people will remind them of that scene from the Wizard of Oz where everyone does the "Jitterbug." Unfortunately for us, Jitter has a completely different meaning in the world of studios, and it's nowhere near as fun as that crazy dance.

Jitter has to do with syncing all of your audio devices together. If they are not synched properly, you get this problem called Jitter which can really mess everything up. In order to understand jitter, we'll have to understand just how we get all our gear running properly.

Since studios have gone digital, the need for synchronizing digital gear has come into focus. Digital audio equipment relies on these things called samples. You may have heard someone say "This session is running at a sample rate of 44.1K." This means that every second, your equipment takes 44,100 samples of your audio. It takes the actual value of the audio signal and converts it into a binary value, storing it somewhere in its memory. All digital devices work with these samples, and with so many samples being taken each second, it is important that everything lines up.

If your equipment does not agree on where to start taking samples, the samples may not line up correctly across all of your gear. This creates jitter. A simple definition of jitter is a mismatch of sampling across multiple digital audio devices. To fix this problem, we introduce a device called a wordclock. The wordclock sends out clocking messages, telling all your devices when to take samples. This matches everything up nicely and eliminates any jitter problems.

Be careful, though. If multiple devices have an internal wordclock, you must set them to listen to your main, external wordclock. If you do not do this, jitter can still be an issue as these devices will ignore your main wordclock and operate on their own. So make sure to implement wordclock properly and switch everything except your main clock to follow the external clock source.

Grounding is not something that usually comes to mind in studio production, but it can be a very big issue if you do not plan for it. Grounding refers to connecting all of your devices to the same electrical ground. When you plug a device into an outlet, it gets sent to ground. If your grounds are all different, you could have some very strange problems, such as hums or buzzes. These can be very distracting when you're trying to mix or record, and you have to think about the proper way to ground things.

Every building has a grounding rod somewhere outside of the building. It's basically a huge rod drilled deep into the ground. If everything gets sent to this same ground, strange issues could arise. Because of this, you'll want a separate ground rod for your studio equipment. Then, you can connect all of your devices to this ground and there will never be any problems of humming or anything. Just make sure you don't connect anything else to this ground (Such as air conditioning), or you'll make your ground line "dirty", which is bad and reintroduces all those hums.

Sometimes, a musician will bring in their own amplifier to record with. When they plug this into your studio output, they might get a very bad hum out of it. Don't worry, you don't have to rewire everything! You can just get what's called a "ground lift" and all your hums will disappear. Basically, a ground lift is a converter for the plug on your device to go from 3 pins down to 2. It eliminates the ground pin, so the device is no longer connected to ground. No more ground, no more ground issues, means no more buzz.

Some high quality amplifiers have a ground lift switch. You can just push this switch to the ground lift setting and you can get by without need for a physical ground lift adapter.

Have you ever had a situation where you've plugged a microphone into your mixer and no matter how much gain you give it, it's never loud enough? You're experiencing a problem called loading. When you load a microphone, your output signal will be very low and you will not be able to get it loud enough to fit well in your mix. Let's look at why this happens and a few ways you could fix it.

Loading occurs when the input resistance of your mixer/preamp is less than ten times more than the output resistance of your microphone. Or, in an equation:Zpre < 10 x Zmicwhere Z represents the total impedance (a fancy word for resistance). Ideally, you want as much input resistance on your mixer and as little output resistance from your mic as possible. This will give you as much voltage into your mixer as possible, thus optimizing your signal for all digital devices. If you're working with tube technology things work a little different.

With tubes and other analog devices, your signal depends on power, not voltage. To achieve the maximum power, you want the output resistance of your microphone to equal the input of your mixer, or:Zmic = ZpreThis will give you the maximum power, and the best signal for analog devices.

But how do you fix having a very resistive microphone? Some mixers have variable resistance switches on their inputs. If you select the biggest setting (the "mic" setting is usually equivalent to 1.5 k-ohms), you'll get the best gain out of your mic. There's no way to reduce the resistance of the microphone, so you'll just have to beef up your preamp.

One trip to your local audio store (Radioshack counts) will show that audio cables aren't cheap. Anything studio quality will cost you an arm and a leg. Lucky for you, making cables is really easy. Here's a simple walk-through on how to make both XLR and TRS (quarter inch) cables.

The first thing you need to do for either connector is buy your materials. Make sure you go with quality connectors, as these cables will last you a long time if you make them properly. Buy some good balanced audio cable, too. After that, you'll have to strip your audio cable. Be careful, though. Audio cable has an outer shielding, then a wire mesh, then two more wires inside separated usually by tightly wound paper. Here's a rough sketch:The wire mesh serves as the ground, the red wire is "hot", and the white wire is "cold" (see the balanced vs unbalanced cable article). To strip the outer stripping, use a razor blade to cut only the rubber (it's too thick for conventional wire-strippers). Pull the outer rubber off to reveal the inner wires. Pull the wire mesh all to one side and twist it into a wire. This will serve as your ground wire. Cut back the paper separator as far as you can. Strip the ends of the red and white wires. "Tin" the ends of the wires. This means you should cover the tips of each wire with a little bit of solder.

Now we have to look at the connectors individually. We'll start with XLR. You'll notice that the connection side has 3 cups sticking out of the connector itself. You'll want to fill up all three cups with solder.

Before we go any further, make sure you have slipped the stress reliever and bottom half of the connector on the cable. If you do not do this now, you will not be able to get it on later. Also make sure to cut a piece of heat-shrink tubing and slide that on last. This will be used to further protect your connections. If you are unsure of what heat-shrink tubing is, it's basically rubber that shrinks to fit when you apply heat.

Then, you have to connect the wires. You should notice each cup has a number associated with it. Number 1 is for ground, 2 is hot, and 3 is cold. Connect your wires appropriately. Simply heat up each cup and let the solder liquify, shove the wire in and let it cool. NEVER blow on a solder connection while it's cooling.

You're done with the hard part! Slide the heat shrink over the connections and heat it up to let it shrink. Attach the connector (this might take a little finangling) and you should be all set.

For a quarter inch connector, you'll see a few pins sticking out from the base. Generally speaking, the one closest to the center is the tip, the next pin is the ring, and the long one one the bottom is the sleeve/shield. Hot goes to tip, cold goes to ring, ground goes to shield. Solder the ground wire directly underneath so that the main cable rests on it, and connect hot and cold whichever way is convenient. Twist on the covering and you're done.

I don't recommend you do this solely based on this text-based description. This can be very confusing the first time you try it, and it's always helpful to have someone show you. This was intended to show you that it's very possible to make your own cables rather easily. If you feel confident that this basic walkthrough is enough to get you making cables, by all means go for it! Trial and error will always show you the right way to do things in the end. If you feel like you need a little more explanation, ask someone who knows about cables and I'm sure they'll be able to give you some pointers.

A few months ago you might have heard debates about "White Space" in regards to the switch in TV broadcasts from analog to digital. Digital signal takes up less bandwidth, thus taking up less white space. But what is it and why does it matter?

White Space is the space we have in which to broadcast signals. Cell phones, tv and radio stations, wireless microphones, garage door openers, and all sorts of other things all use up white space. It is the set of frequency bands that are available for us to use to broadcast in. These frequencies are well beyond the human range of hearing, so we don't hear them but we can use them to communicate over long distances.

By law, wireless microphones, used anywhere from concerts to football games to simple announcements in a large venue, have their own frequency range in which they are allowed to operate. With the switch to all digital TV, the spectrum of wireless mics was going to be opened up to other devices, creating the potential of interference. Many audio engineers were worried that this could be a serious problem, and frantically tested as much as they could before the deadline to switch came about. As it stands, I haven't heard of any issues, but that doesn't mean there aren't any. If you're interested, I'm sure there are plenty of good articles out there about the white space.

With so many cool devices on the market these days, our white space is packed full of signals. Being off just a little bit could cause interference with someone else's signal, which wouldn't be good for anyone. As such, you must get permission to broadcast in a certain frequency band before using any white space of your own.

The title may seem to imply that this is a boring post, but think about it. Cables connect everything in your studio. It's important to know a little bit about your cables, especially when you'll have literally miles of cable in your studio at some point.

An unbalanced cable has two wires inside of it. One serves as the ground, and one has the actual audio signal. This is a pretty simple design, making it cheap and easy to make and implement. Unfortunately, sound can easily find its way into these cables. Only use unbalanced cables over short distances and away from any electrical cords as running unbalanced audio cables for long distances near electrical cords increases the chance of noise to be added to your signal. Noise is bad and sounds awful.

A balanced cable has a third wire inside of it, usually referred to as the "cold" or "neutral" wire. This carries a copy of your audio signal, but inverts the phase. This might seem a little odd, but it allows you to eliminate noise rather easily. Here's a little picture to demonstrate:The red wire is "hot", black is "Ground", and blue is "cold". I drew a sine wave on top of the hot and cold wires. You can see that the sine waves are opposites of each other. Now, let's say sound gets introduced to our signal equally on both wires. If we subtract the signals from each other, we will not only eliminate the sound, but we'll double our signal strength at virtually no cost!This is called subtractive amplification, because you subtract the signals from each other and they actually get louder! If you're a math person, here's the equation that makes this work:A sin (t) - (-A sin (t)) = 2A sin (t)where A is the amplitude and t is time. Since the sound is the same on both, that equation looks like this:A sin (t) - A sin (t) = 0where A this time is the amplitude of the sound signal. This works because the sound is in phase with itself and the audio signal is 180 degrees out of phase with itself. Cool, huh?

Balanced cables are better with noise, but they cost a little more to implement, as you have to continually invert and subtract the signal anytime you transfer it from one location to another. Use balanced cables over longer distances and you'll be pretty safe.

There's a lot of terms that will be flying around once you set foot in a studio, so it's best if you get to know them now. I'll do my best to list all the important ones, but there are quite a few, so I apologize if I miss a couple.

Mic - I hope you know what this is. Mic is short for microphone.

Track - This is what recording engineers refer to a song as. They don't say song, they don't say tune, they say "track." As in, "This track is totally rad." Also, track may refer to a single instrument in the mix. As in, "the bass track could use a boost."

Take - One version of a track. You can (and usually do) have multiple takes of a single track before you find one that you like.

Tracking - The process of recording the raw tracks. No processing is done during the tracking phase. You might get a rough mix just to be able to hear everything properly, but that's about it.

Mixing - What you usually do after tracking. You make all the levels nice, apply EQ and other effects, and get it almost done.

EQ - Equalizer. Attenuating or Boosting certain frequencies of a track or mix.

Mix - The compilation as a whole.

Mastering - This usually comes after the mixing phase. Once you have everything sounding nice, you "master" it. Applying effects to all the tracks at once and making everything sound awesome.

Boom - Unless you're talking about the characteristic of a certain instrument, this refers to a mic stand.

Go-Bo - A moveable partition that you can put pretty much anywhere you want. Helps deflect sound waves.

Effects - Any processing you do to alter a track. Effects include EQ, delay, reverb, distortion, etc.

Reverb - short for Reverberation. Basically, the echoey sounds you get when you talk into a concert hall. Caused by reflections, and simulated with effects processors.

Outboard Gear - any physical device that you can hold in your hand that is not part of your console that can be applied to your mix. Effects processors are included in this.

Console - The device with all the faders and knobs on it. This handles most of your signal processing.

Cans - Headphones that musicians wear while recording.

DI Box - A direct input box. Let's you plug a quarter inch cable into a box, which then has an XLR cable out to plug into your mic panel.

DUT - Device Under Test. This is more technical than the other terms, as it only applies to the design stage of products.

A3F - The female end of an XLR cable.

A3M - The male end of an XLR cable.

MIDI - Musical Instrument Digital Interface. A way to communicate with electronic instruments and sequencers.

Envelopes - Digital controls that define automation of a parameter. For example, a volume envelope defines the volume of a track.

Pots - Short for potentiometers. Basically, a knob that you turn that has an effect. Often used as "pan pots", which control the panning of a track.

Panning - How far left or right of center a sound is perceived.

DVM - Digital Volt Meter. Used for testing/production/repair purposes. Has many different cool features.

Monitor - A really expensive speaker specifically made for use in audio playback in recording studios.

If there's any that you think I missed, comment with a question and I'll go ahead and add it to this list!

Speakers are essentially microphones, except in reverse. Instead of being used to record sound, speakers aim to turn the signal from electrical current back into sound waves. They are actually designed so similarly to microphones that you can actually use a speaker as a microphone (and a microphone as a speaker). This is NOT recommended in any way, I'm just saying it's possible.

Basically, a signal gets sent to your speaker in the form of electrical signals. The speaker then turns these signals into vibrations, which are amplified by the inner workings of the speaker and come out as audio. If you gently touch the cone of a speaker while it's playing music, you can feel that this entire thing vibrates. It's all really rather technical on how it amplifies, and I'm not sure I understand it 100% yet, so I won't post anything on here yet. I promise, though, as soon as I know exactly what happens to the signal while it's inside your speakers I'll publish more on it.

It's important to note here that headphones are the same as speakers, just in a very miniaturized form.

Next up, I'd like to talk a little bit about the lingo used in the professional studio environment.

As promised, we'll now discuss the role of capacitors in condenser mics.

Capacitors operate on a few basic principles. They are comprised of two plates, separated by a very small gap. First, they require a constant to charge to operate properly. The output of a capacitor is based on both the size of the plates and the distance separating them. The equation for the output is as follows: Q = C x V, or Charge = Capacitance x Voltage. Charge can be thought of as the output, and capacitance is the relationship between size of the plates and distance between the plates. Here's a sketch of a capacitor inside a condenser mic:The left plate of the capacitor is attached to the diaphragm of the mic. When the diaphragm vibrates, it moves this plate of the capacitor, changing the capacitance. Let's think back to our equation. The voltage stays constant, being supplied by our 48 volts of phantom power. The change in the capacitance caused by the vibrating diaphragm creates a change in output level, which is the signal that gets sent to our console.

I know it might be a little confusing at first, but go over it a few times in your head. If you still don't quite understand it, Wikipedia's explanation of capacitors can be found here.

Last time we went over basically how a microphone works. Now let's break it down by type.

Before we get started, we need to define a few terms. The first is Transient Response. This is a measurement of how well a microphone reacts to sudden bursts of noise, like a snare hit or a hand clap. Transients are very loud and short, and certain microphones react to transients better than others.

The second term is Proximity Effect. This only occurs in dynamic microphones. If you're speaking into the microphone, and it is less than three inches away from your mouth, you will get an effect called proximity effect. Being less than three inches away from a dynamic microphone creates a very warm, full sound. This is caused by a boost in the bass frequencies. It's actually quite noticeable, so try it when you get a chance. Now let's get started on mic types.

The Dynamic Microphone is the most basic microphone. It operates exactly as outlined in yesterday's post. You talk into the head of the microphone, which vibrates the diaphragm, in turn vibrating the voice coil. This induces current in the mic cable, which gets sent to the console. Very simple and effective. Dynamic microphones are very good for recording vocals. They don't have the best transient response because the voice coil is rather heavy (in comparison to other microphones), but they're very durable, so that's a plus. They also have proximity effect, which is a feature that many musicians desire.

The Condenser Microphone is the next most common mic. These mics require 48 volts of phantom power to operate because they rely on capacitors. To fully understand these microphones, we'll have to delve off into the land of physics for a little bit, so we'll save that for the next post. For now, just know that these mics always need 48 volts of phantom power. Your mixing console will have a phantom power switch somewhere, so make sure that's turned on when you want to use these. They have a much better transient response than dynamic microphones, but they are more fragile. Very loud noises, such as a trumpet wailing on a high note, have a good chance of tearing the diaphragm and thus breaking the microphone. These days it's not such a big deal, but older condensers could have some trouble. Look into the spec sheet for each mic if you're that worried about it.

Ribbon Microphones are like dynamic mics on steroids. They operate on the same basic principle, except the diaphragm on ribbon mics is made differently. It is made from a very thin ribbon of super expensive material. These mics tend to have a great frequency response and better transient response than a regular dynamic mic. Be careful, though, as using a ribbon mic for vocals could spell trouble. They are very sensitive and get dirty very easy. A dirty mic won't work very well, so it's best to keep your distance if you want to sing with one of these bad boys. Also, it doesn't do too well with volume, so be sure to only use it where it won't get blasted, as you can easily tear the ribbon.

One last thing before we leave the general realm of microphones. It's good to have a proper mic handling policy in your studio. One that I've always seen states three places that a mic is allowed to be: In it's case, on a stand, or in your hand. Anywhere else and the mic could get damaged/dirty really easy. So be smart and handle your mics with care.

As I mentioned before, our next post will be a closer look into the condenser mic. Just what makes it so different from those dynamic ones? It's really freaking cool, so brush up on your physics and get ready for our next update.

So you just picked up a shiny new SM-58 and want to know what makes it tick. What really happens when you talk into the top round part? What's that round part really called, anyways? Let's break down the microphone, piece by piece.

A microphone works by turning sound waves into voltage differences, sending them down the cable to your recording device. The top of the microphone is commonly referred to as the head. Most of the head serves to protect the diaphragm. The diaphragm is the part that physically vibrates when you talk into the microphone. The voice coil is attached to the diaphragm. The voice coil is simply a wire wound into a very tight coil. When the diaphragm vibrates (when you talk into the mic), the voice coil also vibrates. This vibration, combined with the magnet, induces a current into the cable which can then be recorded. Here's a basic sketch that might help clarify.There are three main types of microphones we should know about: dynamic, condenser, and ribbon. We'll go over the differences in these three types next.

This is the last post of our basic outline to a recording studio. We've got everything pretty much covered, except for the gear you actually use to do the recordings. That's right, I'm talking about microphones, mic stands, go-bos, amplifiers, headsets... the works. Let's go ahead and dig right in.

The most important parts of the studio are the microphones. Without these, nothing the musicians play will ever get recorded. Ever. So you need to select your microphones wisely. There's always the debate on standards vs. experimental, but in reality it's good to have mics to cover both situations. Here's an example situation to help explain.

A band comes in that wants to play music from the Beatles. Now, the Beatles were very experimental in their day, but many of their techniques are now considered standard techniques. So you need some surefire mics to handle a standard recording setup. Now, there are plenty of brands out there with different price ranges (you almost always get what you pay for). You'll want a good mixture of condenser and dynamic microphones, though (to be covered in our next post). Stay away from ribbon mics for this sort of setup, as they add their own characteristic which the band probably isn't looking for.

Now, picture a different band coming in your studio saying "we want to sound new and different!" Here's where you can try all sorts of new things. Maybe try out that PZM mic which has a cool name and looks kind of funny but you were too afraid to use. Get some ribbon mics on some different things. Have enough mics to accomodate their needs. They may want 8 mics on a piano, for some reason.

You can see that depending on the artists you'd like to bring in, you'll need a different mic selection. We'll go over how different mics work in the next few posts.

This next bit should be fairly obvious, but make sure you have enough mic stands for all of your microphones. Make sure you have different types of stands, too. Straight stands, short booms, long booms, and extra long booms are all essential. If you don't know what those are, look them up. They're pretty self explanatory once you see the different types.

Make sure to have enough headsets for the maximum amount of musicians you expect your studio to accomodate. Without these, musicians will not be able to hear each other very well, and will not perform as well because of it.

One last thing, go-bos. These are partitions on wheels. They allow you to split your one live room up into a few smaller partitions. They block sound waves coming from a specific direction pretty well. All the modern studios have them, so if you can spare the change, get some yourself! You can even make your own pretty effectively.

With that, we end our run down of a studio and start to get much more technical. Next up, we'll talk about the different types of microphones and how signal gets from your instrument to your recordable media.

The first thing that jumps out about the control room is SYMMETRY. An asymmetrical control room will create artificial artifacts during the mixing process, ruining any chance you have of mixing properly.To get a picture of the adverse effects of asymmetrical walls, check out this rough sketch to the right. The black outline are the walls of your control room. You, the listener, is represented by the black dot, and your speakers are the blue dots. The reflections of the sound waves are represented by the different colored lines bouncing all around the room. With this setup, audio travels around the room very unpredictably, causing you to hear the music differently than it actually sounds. Symmetrical walls will make the sound waves travel less randomly, and you'll be able to mix more accurately.

Aside from the walls in your control room, make sure you take all of the noisy equipment out and throw that into the machine room (as stated in a previous post). This will keep your monitoring environment quiet and allow you to concentrate better on the music rather than worry about the hum from all those fans.

Every control room needs three essential things: A mixing console, good monitoring speakers, and chairs for everyone to sit in. I say everyone because there are two very important people that sit in the control room: the Recording Engineer and the Producer. There will also be recording assistants from time to time, but those are the main two. The producer should have his own separate chair, well behind the mixing console (Don't you let them go pushing all those buttons!), and the chief engineer should have a chair directly in front of the console. This allows them to communicate with each other, and gives them both prime view of their domain. Often, the producer's chair is elevated in the back of the room, giving them the feel of power that producers always enjoy.

This next thought gets its own paragraph. Make sure you buy good chairs. There's no need to spend all this money on expensive studio equipment, only to get shoddy, SQUEAKY chairs. Make sure your chairs are squeak-free and everyone will be much happier.

A few words now about your console choice. The number of rooms you have dictates the number of mic inputs you need. If you only have one room, you might only need 16 mic inputs at a time, whereas if you have three rooms, you could easily use up 48 inputs. Be sure to purchase a board that supports your maximum number of mic inputs, or else you'll be losing out on the number of channels available to you.

Another question that comes up concerning the console: digital or analog? Pro Tools or 24-Track Tape? Believe it or not, many studios still use "old fashioned" tape machines in search of that "Analog Warmth" that gets lost in the digital domain. Personally, I'd go with a digital board, recording in to Pro Tools. If you really want that analog sound, you can always record your output from Pro Tools into the Tape Machine, then back into Pro Tools. It won't be exactly the same, but it will help recreate the warmth that many people still desire. Recording strictly to tape means your tracks will be less compatible with other main stream studios, should you ever need to mix anywhere else. Besides, digital boards are really freaking cool.

Analog Effects processors, however, are awesome. People pay big money for outboard gear, and if you're serious about audio, you will too. Plug-ins can recreate many sounds, but these analog devices still work wonders after all these years. You can't quite replicate the sounds you can create with outboard gear, not yet at least. Do your homework, and buy at least a good Equalizer. You won't be sorry.

The next issue has to do with wiring everything together, but I think this is a little too technical for our basic walkthrough. I will definitely post more about this at a later date. For now, I'll just say that there are standards of transferring audio signal that allow you to transfer large amounts of tracks very efficiently. We can get into the specifics later, but you'd be surprised how much data you can fit over one ethernet cable...

Here ends my spiel on the control room. Only a few more things to cover before you'll know the basics to starting your own studio!

So you've got a more detailed sketch of how you want your rooms to look. You got rid of the parallel walls, and started working on reducing sound wave reflections. Now it's time to think about how to keep the sounds in each room separate, and furthermore, how to keep out sounds from outside your studio.

One of the best things you can do to isolate sound is a process called "Floating the Floor." Basically, you put a layer of rubber underneath your floor so that everything is detached. There's a lot more that goes into that, and seek professional guidance if you wish to float the floors in your studio, but it really reduces excess noise. The rubber absorbs most of the vibrations and really helps keep out a low frequency rumble.

If your control room has a window looking into your live room, you can treat the window to prevent leakage. Instead of one flat pane of glass, use two panes, and make sure they aren't parallel with each other (more standing wave issues). This will greatly reduce leakage from the live room, and it looks pretty cool, too.

You can treat your walls with special absorptive insulation that keeps sound both in and out (depending on how you view the situation). Make sure your walls are as big as they can practically be to increase the amount of absorption you can put in. Offset the studs during construction to separate the walls. That way, if one side of the wall is vibrating, the other side isn't dragged along with it. Concrete walls are best, but there are other, cheaper options (such as brick or plasterboard).

These are a few basic things you can do. There are always more ways to isolate your studio, but with these things you should be well on your way. Next up, we'll think a little more about our control room.

Okay, so you've got an idea of how many rooms you want and roughly where you want to put them. Now you've got to design your rooms for the optimal music atmosphere.

The first, and most important thing to make sure of is no parallel surfaces. You might have heard of no parallel walls before, which is true, but the ceiling and floor should not be parallel either. Parallel surfaces have the potential to create standing waves. A standing wave will amplify a certain frequency whenever it is played. This will ruin your recording, so eliminate the parallel walls. This is important for all of the rooms where music is either being recorded or monitored (live room AND control room, for example).

Now that you've got a relatively good shape in mind for your rooms, you've got to think about sound wave reflections. Just because you got rid of the parallel walls doesn't mean you've killed all the bad reflections. Many rooms have natural reverb, and it's all a matter of taste as to how much you want yours to have. To eliminate some reflections, try carpeting your floors and/or walls. These materials will absorb more sound than a tile will. To increase absorption, look for materials that have a high absorption coefficient. This can be looked up online, and I won't got into it here as that gets very technical (a future post will explain absorption coefficients and the like).

Another alternative is to add diffusers. These handy things can be mounted on your walls to help break up sound waves and prevent reflections. The ridges on a diffuser are all different lengths to break up sounds of all different frequencies. As seen below, they come in all different shapes and sizes.One last thing to consider about sound wave reflections: the low frequencies. Lower frequencies don't react to diffusers the same as high frequencies do. To catch the low frequency reflections, use some bass traps. Bass traps are similar to diffusers, but they are made specifically to target the low frequencies. As such, they are much larger, and are actually inserted into the wall. There's also bass traps that are put in the corners of a room which help break up any unwanted reflections, as corners tend to be the worst reflectors.

These few basic principles should be enough to keep you busy until next time. We'll cover more techniques on sound isolation and how to keep outside noise out.

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Chemzoar does not intend to provide complete walkthroughs of everything covered on this website. Before undertaking any projects proposed by this website, be sure to do your own research on top of what's listed here. Chemzoar does not take any responsibility for decisions made based on the content of this site.